Exponential growth in high bandwidth Internet Protocol (“IP”) compliant networks together with new techniques for digitizing analog speech has resulted in significant developments in the field of electronic voice over IP (“VoIP”) communication. Using a common personal computer together with a modem, a user can create a forum in which the user chats with other users thru an IP network. Indeed, a number of vendors including major portal sites provide users with the opportunity to participate in forums.
Despite the promise of modem IP networks, there remain a number of limitations on the bandwidth available for VoIP communication. Uncompressed human speech inherently requires a large bandwidth, a problem that is compounded when multiple people are speaking at once. Various compression techniques have been introduced to address this issue. For example, the International Telecommunications Union (“ITU”) has provided a series of standards for audio compression, known as G series codecs, within the widely adopted H.323 standard.
A codec is a method of compressing digitized voice signals to a compressed digital signal. Each codec compresses digitized voice signals using a particular compression method, such as algebraic-code-excited linear prediction (“ACELP”), multipulse-maximum likelihood quantization (“MP-MLQ”), and low-delay, code excited linear prediction (“LD_CELP”). The result of the operation of a given codec on digitized voice signals is a compressed digital signal produced at a transmitted bit rate that is characteristic of the particular codec. Typically, the transmitted bit rate is constant. For example, within the H.323 standard, the G.711 codec produces a digital signal at a bit rate of 64 kb/s whereas the G.729 codec produces digital signal at a bit rate of S kb/s.
Because a codec compresses digitized voice signals in a predetermined fashion, the quality of the signal produced after decompressing the compressed data is fairly constant and therefore susceptible to measurement. Typically, codecs are rated using a mean opinion score (“MOS”) that ranges from one (poor) to five (excellent). While the use of a codec having a MOS of five is preferable, in practice, such a codec requires a tremendous amount of bandwidth. Thus, compromises are made and standard voice conferences hosted by internet portal sites typically use a codec having a relatively low MOS.
Another shortcoming of standard VoIP platforms, such as those provided by Internet portals, is that they use a single type of codec regardless of the environment in which the VoIP conference is operating. A typical VoIP platform is limited to the use of a lower-speed digital codec, such as G.728 (16 kb/s) or 0.729 (8 kb/s), which have low MOS scores. In fact, the standard VoIP configuration uses a lower-speed digital codec regardless of whether the client is connected by a high bandwidth connection to the network and regardless of network load. Thus, the client of a typical VoIP platform has no option other than to use a relatively low-speed poor quality codec to communicate digital signals to others in the network. This deficiency in the art will tend to become magnified over time, as a growing number of clients switch from the relatively low bandwidth connectivity of a modem to higher speed methods of communication, such as cable modems, ISDN lines, or even T1, T3, or STS-X services.
In view of the above background, it would be highly desirable to provide an improved VoIP environment that is capable of exploiting additional bandwidth capacity when such capacity is present in the VoIP environment.